similar to: problems compiling oh_323 and asterisk

when trying to build asterisk-oh323 I get the following: make[1]: Entering directory `/usr/src/asterisk-oh323-0.5.10/asterisk-driver' gcc -Wall -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declara tions -D_REENTRANT -D_GNU_SOURCE -I/usr/src/asterisk/include -I../wrapper -g -c -o chan_oh323.o chan_oh323.c chan_oh323.c:

Hello, i want to use asterisk as a gateway for H323-Phones. But i cant get it work. I'm using a gatekeeper on another computer. My IP-phone is registered there. Does anybody can sent me an oh323.conf and extension.conf as examples? Thanks in advance Erik Bastian -- NEU : GMX Internet.FreeDSL Ab sofort DSL-Tarif ohne Grundgebühr: http://www.gmx.net/dsl

Hi together, i try to compile astrisk-oh323 like described in the Readme - pwlib V1.6.6 (janus) - openh323 V1.13.5 (janus) with make-patch - asterisk V0.9.0 i got the following error gcc -Wall -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -D_REENTRANT -D_GNU_SOURCE -I/usr/src/redhat/BUILD/asterisk/include -I../wrapper -g -c -o chan_oh323.o chan_oh323.c

I am trying to compile the latest channel drivers Can anyone tell me what is wrong ./check_ver /usr/src/pwlib pwlib ./check_ver /usr/src/openh323 openh323 gcc -shared -Wl,-soname,liboh323wrap.so -o liboh323wrap.so wrapper_misc.o asteriskaudio.o wrapendpoint.o wrapconnection.o wrapper.o wrapcaps.o make[1]: Leaving directory `/usr/src/asterisk-oh323-0.6.2/wrapper' make[1]: Entering directory

hey people, i've installed an asterisk version on my Linux Distribution. Then i wanted to compile the packages for H323 support. The openh323 and pwlib libraries compiled without any error. But at least i got one in /~/asterisk-oh323-0.5-10/asterisk-driver/chan_oh323.c in Line 1128. Too many aruments caused this error. Is anybody able to say why and how can i solve thsi problem? Kindly Regards

This module wont compile can anyone give me any assistance -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040614/03ae433c/attachment.htm

Hi, ALL: I install my oh323 channel driver following steps of http://www.oinko.net/astrecipes/index.php?action=artikel&cat=270174&id=10&artlang=en I works my asterisk well before install the chan_oh323.so. But after I do "make install" the oh_323, my asterisk crash and show me the following message (asterisk -vvvvvvc). Does anyone have any idea about it? What's wrong

I have just setup asterisk on a debian sarge box. I am running Asterisk 1.21 with AMP and chan_capi_cm 0.6.1 using a BT Speedway (AVM Fritz) ISDN card, connected to a BT ISDN2e line. Currently we have 6 extensions (SIP) configured all using CounterPath(Xten) eyebeam softphone. After many hours of Googling I have finally got it all setup and working. We can transfer calls internally and make and

i'm getting Asterisk to dial an h323 call termination service .. right now getting this message: -- Executing Wait("Zap/1-1", "1") in new stack -- Accepting call from '21382890' to 's' on channel 1, span 1 -- Executing Dial("Zap/1-1", "OH323/h323:723@216.52.153.206") in new stack 5:59.330 H323 Cleaner H323

Thanks, but I am still having the same problem -- I made sure to reference the correct passwd file, but WinXP is still forcing me to logon as AGAMEMNON\Guest. Tim -----Original Message----- From: miles123@gmx.de [mailto:miles123@gmx.de] Sent: Monday, May 03, 2004 6:11 AM To: tbbooher@cox.net Subject: SOME Suggestions on [Samba] Can't the very basic stuff to work Hello Tim, I have read

Hi, I start to be a little confused so I am asking to the list. I want to make with * a gateway from PSTN to H323, and to send all incomings call to a predefined IP, which will treat the h323 calls. let's assume that all my incoming numbers starts with 00 here is my extensions [incoming] exten => s,1,Answer exten => _00.,1,Answer exten =>

I got this compiling the new cvs code ... any idea ? Tnx ! gcc -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -g -Iinclude -I../include -D_REENTRANT -D_GNU_SOURCE -O6 -march=i686 -DZAPTEL_OPTIMIZATIONS -DASTERISK_VERSION=\"CVS-04/07/04-11:28:50\" -DINSTALL_PREFIX=\"\" -DASTETCDIR=\"/etc/asterisk\"

I got this compiling the new cvs code ... any idea ? Tnx ! gcc -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -g -Iinclude -I../include -D_REENTRANT -D_GNU_SOURCE -O6 -march=i686 -DZAPTEL_OPTIMIZATIONS -DASTERISK_VERSION=\"CVS-04/07/04-11:28:50\" -DINSTALL_PREFIX=\"\" -DASTETCDIR=\"/etc/asterisk\"

Hello, I have had several instances over the last month of Asterisk freezing, sometimes after 12 hours, sometimes after 8 days. The common elements are that: - all Zap channels lock[hangups don't register and no new calls in or out] - no new in/outbound calls can be made on Zap or SIP channels - people who are still connected to calls can continue to talk - in the CLI interface, you can

I'm following the instruction from Jo?o Amaro from the page http://lists.digium.com/pipermail/asterisk-users/2005-February/090752.html Everything went fine until I run the 'make' command under asterisk-oh323-0.6.5. I got the error message chan_oh323.c:5220: too many arguments to function `ast_channel_register' I have attached the error message. I'm running asterisk CVS

Hello I am trying to compile asterisk-oh323-0.7.0 with pwlib-Janus_patch4 and openh323-Janus_patch4 downloaded from inaccessnetworks so I did this: tar -zxvf openh323-Janus_patch4-src-tar.gz cd openh323 patch -p1 < /root/asterisk-oh323-0.7.0/openh323_1.13.5-make.patch ./configure make opt cd asterisk-oh323-0.7.0 vi Makefile (to set the paths and options according to my system...) NOW I

Hi! I'm looking for a documentation of the .spx file type? I want/need to know how such a .spx file is built up. I want to create a .wav to .spx converter. And additionally a .spx player. I already searched the net and the speex site but I couldn't find anything useful. Maybe I'm missing something? Thanks for your help! Best regards, Peter -- Neu: GMX FreeDSL Komplettanschluss mit

Thanks Paradise, this seems to have worked a treat!!! I commented out the: exten => 110,hint,SIP/110 lines which were in extensions_additional.conf for each sip extension I had. This seems to have stopped the crashes which were previously 3-5 times a day, now: System uptime: 1 day, 18 hours, 10 minutes, 3 seconds Interestingly it had the knock on effect of fixing another problem I had

My fellows, We have Asterisk@home installed and we want to interconnect it with our existing quintum gateways.. any idea how to config that? Your time is very much appreciated.. Cheers, Jessie -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050220/2518797c/attachment.htm

I want the curve to touch the y axis like the curve touches the upper boundary. How can I eliminate the margin between axis and curve on the left side? x1 <- c(1,2,3,4,5) x2 <- c(2,4,6,8,10) mod <- lm (x2~x1) hm <- function (x) (mod$coe[1]+x*mod$coe[2]) plot.new() # ... box() curve (hm,lty=1,add=T,xaxs="i",yaxs="i") (R 2.8.1) -- Sensationsangebot verl?ngert: GMX